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Voice over IP (VoIP)


Overview

 Why VoIP?  Voice Codec  Signaling & Control Protocols  Quality of Service  Design a VoIP network

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Why VOIP ?  Cost reduction  Toll by-pass

 WAN Cost Reduction 

Operational Improvement  Common network infrastructure  Simplification of Routing Administration

Business Tool Integration  Voice mail, email and fax mail integration  Web + Call  Mobility using IP

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Yesterday’s Network

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Converged Network

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Components of VOIP  Coding & Decoding of Analog Voice  Analog-to-Digital and Digital-to-Analog conversions  Compression  Signaling  Call setup & tear down  Resource & coding negotiation  Transport of Bearer Traffic  Voice packet transmission  Routing  Support of quality of service  Numbering  Phone number, IP address

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What Protocols are Required ?

 Signaling Protocol: To establish presence, locate user, set up, modify and tear down sessions.

 Media Transport Protocols: To transmit packetized audio/video signal.

 Supporting Protocols: Gateway Location, QoS, AAA, Address Translation, etc.

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VOIP Protocols  H.323:  ITU-T standard, latest version v4  Peer-to-peer protocol that supports terminals communicating over packet based networks

 SIP:  IETF standard, RFC 3261  Peer-to-peer protocol for initiation, modification termination of communication sessions between users

 MGCP:  ITU-T and IETF collaboration, RFC 3435  Master/slave protocol for media gateway controller to control media gateway.

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VOIP Protocol Stacks

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VOIP Using H.323  ITU-T standard, latest version v4  Peer-to-peer protocol that supports terminals communicating over packet based networks  H.323 includes  Call signaling H.225  Media control H.245  Audio coding G.711, G.722, G.723, G.728, G.729  Video coding H.261, H263  Data sharing T.120  Media transport RTP, RTCP

 Powerful for video-conferencing  Widely deployed

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H.323 Components

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Components Defined in H.323  H.323 defines four major components for a packet network based multimedia communication  H.323 Terminal: Client end points that provides real-time communications with other H.323 entities. Functions performed include (1) signaling and control, (2) real-time communications, and (3) codec.  Gateway: Provides the connection path between the packet switched network and the switched circuit network.  Gatekeeper: Performs address translation, admission control, bandwidth control, zone management, call control signaling, call authorization, bandwidth management, call management.  Multipoint control unit (MCU): Supports conferencing between three or more endpoints. MCU consists of multipoint controller (MC) and multipoint processor (MP).  The collection of all terminals, gateways, and MCUs managed by a single gatekeeper is called a zone.

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SIP Registration

SIP Registration Establishes presence of user with an address (eg.dave@abc.com) Binds this address to users current location (199.147.77.23)

Register sip:abc.com SIP/2.0 From sip:dave@abc.com Contact <sip:199.147.67.23> Expires 3600 1

dave@ 199.147.67.23

Location sever

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SIP register SIP / 2.0 200 OK

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SIP Operation with Proxy Server Location Server

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OK 200 VIA: dave@abc.com FROM: sip:dave@abc.com TO: sip:john@xyz.com Call-id: 1234@abc.com

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Where is john?

INVITE sip:john@xyz.com VIA: dave@abc.com FROM: sip:dave@abc.com TO: sip:john@xyz.com Call-ID: 1234@abc.com

john@195.127.75.123

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INVITE sip:john@195.127.75.123 VIA: proxy@internet.com VIA: dave@abc.com FROM: sip:dave@abc.com TO: sip:john@xyz.com Call-id: 1234@abc.com

5 Proxy Server

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ACK sip:john@xyz.com VIA: dave@abc.com FROM: sip:dave@abc.com TO: sip:john@xyz.com Call-id: 1234@abc.com

Media Streams 14

OK 200 VIA: proxy@internet.com VIA: dave@abc.com FROM: sip:dave@abc.com TO: sip:john@xyz.com Call-id: 1234@abc.com


SIP operation with Redirect Server

Location Server

302 moved temporarily Contact john@def.com ACK john@def.com

Where is john

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john@def. com

INVITE sip:john@xyz.com VIA dave @abc.com 1 From sip:dave@abc.com To sip:john@xyz.com Call ID 1234@abc.com

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Redirect Server 4 INVITE sip:john@def.com VIA dave@abc.com To sip:john@def.com Call ID 1234@abc.com

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OK 200

8 ACK john@def.com Media Stream 9

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Session Initiation Protocol (SIP)  Peer-to-peer protocol for initiation, modification termination of communication sessions between users  Two components: user agents and network servers User agents: client end-system applications  User-agent client (UAC): originate calls  User-agent server (UAS): listen for incoming calls

 Network servers:  Proxy server: relay calls, act as both client and server  Redirect server: redirect calls to other servers  Registrar: accept user registration

 Simple text based messages

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SIP Messages  SIP is a text-based protocol with message syntax and header fields identical to HTTP  Message header includes:  General header  Entity header

 Two kinds of messages:  Requests initiated by client  Responses returned by server

 Methods defined in RFC 2543:  INVITE: to invite the server to participate in a session  ACK: to accept the INVITE to participate  OPTIONS: to inquire capability  BYE: to terminate a session  CANCEL: to cancel any in-progress request  REGISTER: a client to register location information with a server

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SIP Messages

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voip  

voip setup

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